Service level agreements based on objective voice quality testing for voice over IP (VOIP) networks

ABSTRACT

An objective, service-level specific voice call listening quality test scheme for a Voice Over IP (VOIP) network is presented. Test probes are deployed along the border of the VOIP network. Each test probe is capable of placing calls over the VOIP network to the other test probes at different levels of service and measuring call quality using an objective measurement algorithm such as PAMS or PSQM. The measurement results are collected on an ongoing basis to obtain information on the VOIP network&#39;s voice call quality. The information is compared to thresholds to measure performance against Service Level Agreement guarantees.

BACKGROUND OF THE INVENTION

[0001] The invention relates to voice call quality testing.

[0002] Packet-based networks, in particular, Voice Over IP (VOIP)networks, are rapidly emerging as a viable alternative to traditionaltelephony (that is, circuit switched networks). VOIP is viewed as anattractive option for voice transport in that it allows live voiceconversations to be integrated with existing IP data and imageapplications. To be a truly competitive alternative, VOIP must emulatethe performance of traditional telephony and do so using a protocol thatwas optimized for data traffic. The characteristics of data traffic arequite different from those of voice traffic, however.

[0003] Unlike data traffic, voice traffic is extremely intolerant ofdelay and delay variation (or “jitter”), as well as packet loss. Muchwork has been done in the area of packet delivery to provide end-to-endQuality of Service (QoS). Service level agreements (SLAs) for VOIP, likethose for conventional data IP networks, therefore tend to be based onconventional data network metrics, that is, guaranteed service levelsare expressed solely in terms of packet level performance, e.g., packetloss, delay, jitter.

[0004] Another important aspect of voice communications quality that isnot reflected in the data network metrics, however, relates to the soundof a voice call from the perspective of the listener. Standardizedtechniques exist for measuring this aspect of voice quality. Typically,to support voice communications, VOIP networks encode the audio andformat the encoded audio into packets for transport using an IPprotocol. Consequently, the results of these voice quality tests aregreatly affected by choice of speech coding techniques.

[0005] One approach utilizes a standardized ranking system called theMean Opinion Score (MOS). The MOS system uses a five-point scale:excellent-5; good-4; fair-3; poor-2; and bad-1. A level of qualityindicated by a score of 4 is considered to be comparable to “tollquality”. A number of people listen to a selection of voice samples orparticipate in conversations, with the speech being coded by using thetechnique to be evaluated. They rank each of the samples orconversations according to the five-point scale and a mean score iscalculated to give the MOS. Although the applicable ITU standard, ITU-TP.800, makes a number of recommendations regarding the selection ofparticipants, the test environment, explanations to listeners, analysisof results, etc., because this type of voice quality testing issubjective, results tend to vary from test to test.

[0006] Algorithms for performing a more objective voice quality testhave been developed as well. These objective techniques for testingvoice transmissions in packet-based networks include an ITU standardbased algorithm known as Perceptual Speech Quality Measurement (PSQM)and Perceptual Analysis Measurement System (PAMS), which was developedby British Telecom. Both of these perceptual test algorithms evaluatewhether a particular voice transmission is distorted from theperspective of a human listener.

[0007] PSQM takes a “clean” voice sample and compares it to apotentially distorted version, that is, a transmitted version, using acomplex weighting that takes into account what is perceptually importantto a human listener, for example, the physiology of the human ear andcognitive factors related to what human listeners are likely to notice.PSQM provides a relative score (on a scale of 1 to 15, with 1corresponding to the highest score and 15 corresponding to the lowestscore) that indicates how different the distorted signal is with respectto the reference from the perspective of the human listener.

[0008] PAMS is based on a perceptual model similar to that of PSQM andshares with PSQM the purpose of providing a repeatable, objective meansfor measuring voice quality, but uses a different signal processingmodel than PSQM and produces a different type of score as well. Thescoring provided by PAMS includes a “listening quality” score and a“listening effort” score, both of which correlate to the MOS scores andare on the same scale of 1 to 5.

SUMMARY OF THE INVENTION

[0009] In one aspect of the invention, providing service for use in aVoice Over Internet Protocol (VOIP) network environment includesselecting a service level and measuring voice call listening qualityaccording to the selected service level for voice calls transmittedacross a VOIP network to produce voice call listening quality metricvalues.

[0010] In another aspect of the invention, a computer program productresiding on a computer readable medium for providing service for use ina Voice Over Internet Protocol (VOIP) network environment comprisesinstructions for causing a computer to: associate service levels withphone numbers; and, in response to a test voice call directed to one ofthe phone numbers, cause the test voice call to be transferred over theVOIP network to a destination corresponding to such phone number at theassociated service level and causing a voice call listening quality tobe measured for the associated service level to produce a voice calllistening quality metric value.

[0011] Particular implementations of the invention may provide one ormore of the following advantages. The invention allows a networkoperator or service provider to provide a customer with an SLA that isbased on voice call quality metrics, in particular, voice call listeningquality metrics (such as MOS rankings), which reflect voice call qualityas perceived by a human listener, as well as packet-based metrics. Thevoice quality tests are objective, repeatable tests and can therefore beimplemented in an automated, production environment to enforce SLAs. Theinvention also enables the voice call listening quality metrics to beobtained for different service levels according to coding scheme and/orprotocol.

[0012] Other features and advantages of the invention will be apparentfrom the following detailed description and from the claims.

BRIEF DESCRIPTION OF THE DRAWINGS

[0013]FIG. 1 is a block diagram illustrating a Voice Over IP (VOIP)network voice call listening quality test topology.

[0014]FIG. 2 is a block diagram of an exemplary VOIP-telephony networkthat employs the test topology of FIG. 1.

[0015]FIG. 3 is a detailed illustration of an exemplary VOIP Point ofPresence (POP).

[0016]FIG. 4 is an illustration of a VOIP gateway configurationsupporting voice call listening quality testing for a VOIP network.

[0017]FIG. 5 is an alternative embodiment of the test topology of FIG.1.

[0018]FIG. 6 is a VOIP-telephony network that employs the test topologyof FIG. 5.

DETAILED DESCRIPTION

[0019] Referring to FIG. 1, an exemplary voice quality network testtopology 10 includes a packet-based network shown as an IP network 12that transports voice traffic. The network test topology 10 furtherincludes voice quality test probes 14 a and 14 b, which are coupled togateways 16 a and 16 b, respectively. The gateways 16 a and 16 b eachare connected to the IP network 12 and provide translation servicesbetween protocols of the IP network and a conventional telephonynetwork, such as a Public Switched Telephone Network (or “PSTN”). Thetest probe 14 a is connected to the gateway 16 a by a first telephonytransmission line, shown as an ISDN line 18 a (e.g., E1 or T1), and usesan ISDN Primary Rate Interface (PRI) service. The test probe 14 b isconnected to the gateway 16 b over a second telephony transmission line18 b, also shown as an ISDN line 20 b supporting ISDN-PRI service. OtherPSTN physical and signaling interfaces can be used. For example, thelines 20 a, 20 b may be ISDN-BRI or CAS T1/E1 lines. Alternatively, thelines can be implemented as analog FXO wires. Because the IP network 12transports voice traffic, it is also referred to as a Voice Over IP(VOIP) network and IP communication devices coupled to the VOIP network12 either directly (such as the gateways 16 a, 16 b) or indirectly(through another IP communications device) are also referred to as VOIPcommunications devices.

[0020] The test probes 14 a, 14 b store a sample or reference voice file22 a, 22 b, respectively, for test purposes. The reference voice file 22is formatted as a typical audio file, e.g., RIFF WAVE “★.WAV” file (asshown), or some other audio format. The test probes 14 a, 14 b alsostore a software algorithm implementing a perceptual or voice calllistening quality test model. In one embodiment, the software algorithmis the Perceptual Analysis Measurement System (PAMS) algorithm. Otherobjective, repeatable voice call listening quality test algorithms,e.g., Perceptual Speech Quality Measurement (PQMS), can also be used.Although one of the two test scores produced by PAMS is known as“listening quality”, the term “voice call listening quality” as usedherein refers to the quality measured by any perceptual voice call testtechnique, such as PAMS (and therefore encompasses both the PAMS“listening quality” as well as the PAMS “listening” effort”) or PQMS.The test probes 14 a, 14 b can store any type of reference voice file,but the voice file stored on both of the test probes 14 as the referencevoice file must be identical in order for the PAMS (or other similar)voice call listening quality testing to work correctly. Preferably, foroptimum test results, the voice file should include voice samplesrepresentative of many different types of voice activity.

[0021] In operation, the test probes 14 transmit and receive thereference voice files (test.WAV files 22 a, 22 b) over the speech pathwithin the VOIP network. One test probe acts as a resource to transmitthe file. A second test probe acts as a resource to receive the filetransmitted by the first test probe and perform the PAMS algorithm. Forexample, the test probe 14 a, serving as a “call initiator”, dials atelephone number corresponding to the other test probe, the test probe14 b (acting as a “call responder”). The VOIP gateway 16 a directs thecall over the VOIP network 12 to the VOIP gateway 16 b, which sends thecall to the test probe 14 b. The test probe 14 b answers the call byplaying back to the caller, that is, test probe 14 a, the storedreference voice file. The test probe 14 a records the played voice fileas it “listens” and analyzes the voice quality of the recorded voicefile using the PAMS algorithm. The PAMS algorithm compares the recordedvoice file to the reference voice file stored by the test probe 14 a anddetermines a difference between the files. When the analysis iscomplete, the test probe 14 a translates the difference into a PAMSscore (actually, a two-part score having separate scores for listeningquality and listening effort) for the voice quality of the call. It willbe understood that the same process can be used in the reversedirection, that is, when the test probe 14 b acts as the call initiatorand the test probe 14 a acts as the call responder. The test probes 14may be configured to allow a test administrator to view the resultingscores graphically or in some other form.

[0022] Preferably, the voice call listening quality test is performedfor each level of service as determined by the type of codec (i.e.,coder/decoder) that is used by the VOIP communications device that isperforming the voice encoding and decoding operations. In the networkshown in FIG. 1, the gateways 16 implement one or more coding schemes tosupport voice encoding/decoding.

[0023] Types of codecs include, but need not be limited to, thefollowing: waveform codecs, source codecs and hybrid codecs. Withwaveform codecs, an incoming voice signal is sampled, coded and thecoded samples converted to quantized values, which are used toreconstruct the original voice signal. Waveform codecs produce highquality sound but consume a significant amount of bandwidth. Sourcecodecs try to match an incoming voice signal to a mathematical model ofspeech generation. That model is used to reconstruct the original voicesignal. The source codec operates at low bit rates but tends to producepoor quality sound. Hybrid codecs use some amount of waveform matchingas well as knowledge of how the original sound was generated. They tendto provide fairly good quality at lower bit rates than waveform codecs.

[0024] The G.711 Pulse-Code Modulation (PCM) coding technique is awaveform codec and is one of the most common coding techniques that areused. It is the codec of choice for circuit-switched telephone networks.Other PCM waveform codecs include G.726, which offers AdaptiveDifferential PCM (ADPCM) coded speech. Lower bandwidth, hybrid codecsinclude the G.723 and G.729 codecs.

[0025] The choice of codec is a major factor in high voice quality andvoice quality test scores will vary with codec selection. The goal ofany network operator or service provider is to offer “toll quality”voice, where toll quality voice relates to a MOS of 4.0 or better. Thetask of selecting the best codec for a given network is a matter ofbalancing quality with bandwidth consumption. Thus, it is desirable totest different codecs in an environment that closely matches expectednetwork conditions.

[0026] Still referring to FIG. 1, the test topology 12 tests for threedifferent coding techniques, G.711, G.723 and G.729, although othercoding schemes can be used in addition to, or lieu of, any one or moreof these techniques. That is, each of the gateways 16 is capable ofencoding and decoding in accordance with these different codingtechniques. Consequently, there is a different phone number for eachtest probe and service level combination.

[0027] Optionally, in addition to measuring voice call quality on aper-codec basis, test probes 14 can test voice call quality on aper-protocol basis as well. That is, it is possible to further associateeach test probe phone number and service level combination with eachprotocol supported by the VOIP communications devices 16. For example,commercially available gateways support such signaling protocols asH.323 and Session Initiation Protocol (SIP), as well as Media GatewayControl Protocol (MGCP). Other existing protocols, such as media gatewaycontrol protocol (MEGACO/H.248), as well as other protocols, may besupported as well. The protocol-specific testing could measure, forexample, voice call setup and tear-down times.

[0028] For example, to illustrate service level selection based on bothcodec and IP signaling protocol, each of the test probes 14 may beassigned four unique telephone numbers. The four unique phone numbersinclude: a first unique phone for a service level associated with theuse of G.711 coding and H.323 signaling; a second unique phone for aservice level associated with the use of G.723 coding and H.323signaling; and third unique phone number for a service level associatedwith the use of G.711 coding and SIP signaling; and a fourth uniquephone number for a service level associated with the use of G.723 codingand SIP signaling. To place a call to the test probe 14 b, the testprobe 14 a initiates a call to one of the four unique phone numbers forthe test probe 14 b with the gateway 16 a. Gateway 16 a is configuredwith resources to perform both types of coding and signaling, butselects the appropriate coding for the call to the test probe 14 b andcall signaling to establish a connection with the gateway 16 b based onthe phone number. Typically, the gateways include a configuration tablewhich stores the called phone numbers with associated service levelinformation for look-up, as will be described in further detail below.Thus, the gateway 16 a determines from the service level informationassociated with the called phone number (for test probe 14 b) how theaudio is to be encoded and how the data connection is to be established.When the connection between gateway 16 a and 16 b is established, thegateway 16 b is able to detect the type of coding in use and allocate anappropriate coding resource to encode the audio when it is receives thevoice file transmission from the test probe 14 b.

[0029] Although not shown, it will be understood that the test probes 14further include the necessary hardware and software required to supportapplicable network layer protocols. In addition, and specifically insupport of voice call quality testing, each test probe 14 includes acall generator. In one embodiment, the call generator provides acomplete H.323 implementation package that is capable of initiating andresponding to calls. The package thus simulates an H.323 terminalgenerating calls with (or without) a VOIP gateway, as well as openinglogical channels and transmitting RTP voice packets. Other VOIPprotocols, such as SIP and MGCP (as discussed above), can be usedinstead of or in addition to H.323.

[0030] As indicated above, the test probes 14 are configured to performautomated voice quality measurements on a voice transmission and producea score based on those measurements. This testing may be performed in alaboratory environment to simulate conditions of an operating network,or as part of the actual network operation, as will be described withreference to FIG. 2.

[0031]FIG. 2 depicts an exemplary telephony-VOIP network 30. Theexemplary network 30 illustrates how the test topology of FIG. 1 isadapted for use in a production environment. The network 30 deploys anumber of test probes like the test probes 14 (FIG. 1), indicated byreference numerals 14 a through 14 f, at different points along theborder of the VOIP network 12. Although not shown in the figure, each ofthe test probes in configured with a copy of a reference voice file, asdescribed earlier with reference to the reference voice files 22 ofFIG. 1. Each of test probes 14 a through 14 d is connected to arespective one of gateways 16 a through 16 d. Also connected to thegateway 16 c is a server 31. Each of the gateways 16 a through 16 d isconnected to the VOIP network 12 and a respective one of PSTNs 32 athrough 32 d. In addition to the gateways 16, VOIP communicationsdevices include a VOIP server 34 and a VOIP telephone 36. The test probe14 e is connected to the VOIP server 34 and the test probe 14 f isconnected to the VOIP telephone 36. Each of the test probes 14 iscontrolled to generate test calls to others of the test probes 14 overthe VOIP network 12, perform PAMS testing on the voice files played backin response to the test calls, as well as play a reference voice filewhen acting as a recipient of a test call, much in the same manner aswas described for the two test probes shown in FIG. 1.

[0032] The test probes 14 attach to VOIP communications devices, such asdevices 16, 34 and 36, through digital or analog circuits. In addition,or alternatively, test probes can be deployed at other locations fordifferent types of voice quality (and possibly packet-based) testcoverage. For example, active test probes can connect to PSTNs throughtelephony interfaces for end-to-end voice quality testing, like testprobes (TP2) 40 a, 40 b, which are connected to PSTNS 32 c and 32 d,respectively, or can connect directly to the VOIP network 12 through anIP interface and appear to the VOIP network 12 as another gateway, orsome other VOIP communications device. The latter configuration, anexample of which is illustrated by test probe TP1 38, only tests theVOIP network, not the gateway, however. A “passive” test probe may beconnected between the gateways 16 (or other VOIP communications devices)and the VOIP network 12 to produce information for all of the voicecalls it sees, in particular, packet-based and voice quality informationfor all of the voice calls and PAMS data for probe-generated voicecalls. Test probe (TP3) 42, which is coupled between the gateway 16 band the VOIP network 12, is an example of a passive test probe.

[0033] All of the test probes store a copy of the same reference voicefile and have the capability to generate PAMS scores for test calltraffic. All but passive test probes can generate and answer test callsin the manner described above. Unlike the other probes, the passiveprobe TP3 42 sees all voice calls, including test calls. It is able toidentify a test call by a caller's IP address or the called phonenumber. Once a test call is detected, the passive probe can extract theaudio from a test call and apply a PAMS test to it. Consequently, thepassive probe provides a test result for a network location intermediatethe call source and destination points. A combination of end-to-endscores, border-to-end (or border) scores, as well as a passive probescores, therefore enables a test administrator to isolate a networktrouble spot.

[0034] In addition to PAMS testing, the end-to-end voice quality testingtest probes 40 also include software to support other active testmeasurements for signaling and voice quality, including: post dialingdelay; post gateway answer delay; background noise; audio level;insertion loss; round trip delay; echo and DTMF integrity.

[0035] The VOIP communications device test probes, such as test probe38, can perform, in addition to the PAMS tests, the following signalingand voice quality tests: Q.931 setup time; RTP setup time; backgroundnoise; audio level; and insertion loss. The software can also simulatevarious packet-based impairments (e.g., jitter, total packet loss,packet loss burst, etc.) and assess their effects on the VOIP and theVOIP communications devices, e.g., the VOIP gateways 16. The voicequality measurements by the VOIP communications device test probes canbe end-to-edge or edge-to-edge.

[0036] In addition to performing voice quality tests on PAMS-generatedtraffic, passive test probes, such as the test probe 42 can support, forall live traffic monitored on the VOIP network 12, the following activetest measurements: call statistics; setup time; jitter per RTP stream;and packet loss (and packet loss burst) per RTP stream.

[0037] Thus, the deployment of the various types of probes throughout anetwork such as network 30 provides for a network-wide monitoringsystem. The different types of test probes, that is, the test probes 14,38, 40 and 42, and the manager 44 shown in FIG. 2 may be implementedusing commercially available hardware and software, for example, usingthe various components of the Omni-Q Voice Quality Management Systemmanufactured by and available from RADCOM.

[0038] Still referring to FIG. 2, also connected to the VOIP network 12is a management server (or manager) 44. All test probes in the network30 are configured and controlled by the manager 44. The managerconfigures the test probe properties and test call generation schedules,as well as establishes alarms or thresholds to ensure delivery ofservice in accordance with Service Level Agreements (SLAs). It polls theprobes on a periodic basis to gather test results, consolidates the testresults for the entire network and stores the consolidated informationin a database for analysis, reporting and historical trending.

[0039] In one exemplary commercial setting, the VOIP networkinfrastructure of the network 30 is maintained by a VOIP networkoperator (or wholesale service provider) and made available to that VOIPnetwork operator's customers, e.g., retail service providers who use theinfrastructure and related services of the VOIP network operator toprovide services to end-users. The network operator manages the VOIPnetwork infrastructure and related services using the manager 44. In alarge-scale operation, the network operator supports a large number ofVOIP Points of Presence (VOIP POPs), an example of which is indicated indashed lines as VOIP POP 46, in different geographic regions forcoverage of a larger territory, e.g., national level coverage. Each VOIPPOP provides a point of entry to and termination from the VOIP networkbackbone. The network operator supports outbound call completion to thePSTNs for calls placed over the VOIP network and inbound call deliveryfor calls initiated on the PSTN for delivery over the VOIP network. Anexample of such an inbound service may be delivery of calls fromend-users to a customer's unified messaging server. To support such aservice, the network operator provides a block of local Direct InwardDial (DID) numbers to the customer so the customer has local accesscapability in a particular geographic market. A call from an end-user toone of the DID numbers results in a call passing through a gatewayassociated with that number to the customer's server (via a gatewayhosting that customer). Examples of other inbound call services include,for example, conference call bridging, call delivery to call centers andcall waiting servers. The network operator provides service levelagreements based on voice call quality, which includes voice calllistening quality metrics (i.e., test score values) as well aspacket-based metrics, as described above.

[0040] Referring to FIG. 3, a network 30′ having an exemplary VOIP POP46′ that supports inbound service delivery and service level based voicecall quality testing is shown in detail. In this example, the VOIP POP46′ includes multiple gateways 16-1, 16-2, . . . 16-8 to supportdifferent metropolitan areas, including San Francisco, Los Angeles andSan Diego (as shown). The closest Internet access point is located inLos Angeles, so traffic from San Francisco and San Diego is directed toLos Angeles, where it is passed to an Internet backbone router 50. Eachgateway 16 in each metropolitan area has multiple connections 52 tosupport customer traffic and at least one gateway 16 in each areasupports a single connection 54 for connecting to one of three testprobes 14-1, 14-2 and 14-3. The connections 52 in each area connect arespective PSTN to that area's gateways 16. Thus, the two gateways inSan Francisco, the gateways, 16-1 and 16-2, are connected to a PSTN inSan Francisco (SF PSTN), 32-1, and the gateway 16-1 is also connected tothe test probe 14-1. The four gateways in Los Angeles, the gateways,16-3, 16-4, 16-5 and 16-6, are connected to a PSTN in LA (LA PSTN),32-2, and the gateway 16-6 is also connected to the test probe 14-2.With respect to San Diego, the gateways 16-7 and 16-8 are connected to aPSTN in San Diego (SD PSTN), 32-2, and the gateway 16-8 is alsoconnected to the test probe 14-3.

[0041] In the illustrated example, and as discussed above, the VOIPnetwork operator enables customers to deploy their services on anational level without having to make an investment in networkinfrastructure. Blocks of local DID numbers are made available tocustomers for use by end-users. In this example, it is assumed that oneof the San Francisco gateways 16-1, 16-2 is configured to handle phonenumbers having an area code “408” and a three-digit exchange of “123”,followed by a four-digit number in the range of 1000-3000. Twocustomers, customers A and B, wish to service users in the San Franciscoarea and thus require blocks of local DIDs to give to customers in thatarea. The identified gateway therefore serves as a central office andall the inbound traffic is aggregated to a single connection point.

[0042] In general, and referring to FIG. 4, the gateways 16 maintain aconfiguration table 60 that includes, for each of a plurality ofprofiles 61, a phone number 62, an associated service level 64 androuting information 66. At minimum, the service level 64 indicates thecoding scheme to be used by the VOIP communications devices responsiblefor establishing the IP data path over which the voice transmission isto occur. The routing information 66 identifies by IP address a hostinggateway or server to which inbound traffic associated with the phonenumber 62 is to be directed.

[0043] As an example, and thus intended for illustrative purposes only,the second, third and fourth table entries are populated with data tosupport configurations for inbound VOIP calls for the two customers,customer “A” and customer “B”, and SF test probe 14-1 via one of thegateways 16-1, 16-2 (FIG. 3). Numbers in the 1000-1999 range areallocated to customer A, who has requested G.711 service. The numbers inthe range of 2000-2999 are allocated to customer B, who has requestedG.723 service. The “3000” number is assigned to the test probe 14-1 fora specific service level, for example, G.711.

[0044] Referring to FIGS. 3 and 4, when one of the San Franciscogateways, say, 16-1, configured with the configuration data shown in theconfiguration 60, detects an in-coming call from the PSTN 32-1, itparses the configuration 60 to match the incoming phone number to any ofthe stored phone numbers. If the gateway 16-1 determines that the dialednumber belongs to a particular customer such as customer A, it selectsas a service level the service level requested by customer A, that is,the G.711 service level (specified by the service level field 64). Itroutes the call to the gateway identified by the routing informationfield 66 for hosting customer A, for example, the gateway 16 c (showncoupled to the server 31, representing customer A), using the G.711service level. A call belonging to customer B is handled in much thesame way, but according to the specific configuration information forcustomer B. That is, the call is routed to a hosting gateway or a serverfor customer B, shown in the figure as the server 34, using the G.723service. If the dialed number is 408-123-3000, the gateway 16-1determines from the configuration 60 that the call is a test call to beplaced to a test probe, for example, the test probe 14 d, using G.711service. At this point, the process is as described above with referenceto FIG. 1.

[0045] As already indicated, the manager 44 is operated by the IPservices provider to control the test probes deployed throughout theVOIP network 12. The manager 44 determines the frequency with which thetest probes make test calls and schedules the test probes to generateand receive the test calls. It polls the test probes for test results(MOS scores) and is able to process the raw data for reporting, networkrepair/enhancements, and so forth. For example, the manager 44 candetermine an average score from all of the test calls for each servicelevel, that is, G.711, G.723 and G.729, during a given time period(e.g., on a monthly basis) and compare that average performance metricto a guarantee provided by an SLA between the VOIP network operator(service provider) and a service subscriber (such as the hypotheticalcustomers A and B in the above-described example).

[0046] Other embodiments of the voice quality testing topology andVOIP-telephony network of FIGS. 1 and 2, respectively, are contemplated.For example, and referring to FIG. 5, a voice quality test networktopology 70 includes the VOIP network 12 and coupled to the VOIP network12 are a test probe 72 and a VOIP communications device 74. The testprobe includes a reference voice file 76 a and the VOIP communicationsdevice 74 has an embedded reference voice file 76 b. The test probe 72can be functionally identical to the test probe 38 shown in FIG. 2. Thetest probe 72 and the VOIP communications device 76 each include asuitable interface to allow direct connection to the VOIP network 12over an appropriate VOIP network connection 80, e.g., an Ethernetconnection. Thus, in contrast to the topology shown in FIG. 1, thetopology 70 eliminates a second test by providing a voice file in a VOIPcommunications device having a codec to be evaluated. The VOIPcommunications device 76 can be any VOIP communications device thatperforms speech encoding/decoding in a particular VOIP networkenvironment, for example, a gateway, a server, a telephone (like thegateway 16, VOIP server 34 and VOIP telephone, respectively, from FIG.2), or any other VOIP communications device.

[0047] In addition to the voice file 76, the VOIP communications devicemust also have a voice quality test support module 82 to enable thedevice to answer a test call by playing the voice file. As indicated inthe figure, the module 82 can be implemented as an Interactive VoiceResponse (IVR) unit as is well known in the art. At present, many VOIPcommunications devices already include an IVR unit to support otherfunctions, such as interactive call processing functions. In thoseinstances, the voice call listening quality test scheme is able toexploit a device's inherent interactive voice response (IVR) capability.Alternatively, if the device does not include IVR functionality, thenthe device requires a script enabling that device to recognize a specialphone number and understand that a call to that special phone number isto be answered by playing the embedded voice sample file.

[0048]FIG. 6 illustrates an alternative VOIP-telephony network 90generally configured in a similar configuration to that of the network30 of FIG. 2, but modified so as to require only one test probe toproduce PAMS scores for the voice call quality between the test probeand various connection points around the VOIP network 12 according tothe test topology 70 shown in FIG. 5. That is, and with reference toFIGS. 2 and 6, the test probes 14 are eliminated by embedding voicefiles (VF) 92 in the VOIP communications devices to which the testprobes 14 were coupled, that is, the gateways 16, VOIP server 34 andVOIP phone 36, shown with the embedded voice files in FIG. 6 as gateways16′, VOIP server 34′ and VOIP phone 36′. The test probe 38 isresponsible for generating calls to the gateways 16 a′, 16 b′, 16 c′ and16 d′, as well as device 34′ and 36′ (on a scheduled basis under thecontrol of the manager 44) and generating PAMS scores from the responses(that is, the playback of stored reference voice files) by the calleddevices 16′, 34′ and 36′. The functionality of the manager 44 is asearlier described with respect to FIG. 2.

[0049] Additions, subtractions, and other modifications of the describedembodiments of the invention will be apparent to those practiced in thisfield and are within the scope of the following claims. The test schemeand topology can be adapted to accommodate other different physical andsignaling protocols. For example, the test probe 40 could be connectedto an SS7 network, or the test probe 14 could connect to an SS7 or CASgateway interface. Moreover, the packet-based network 12 need not be anIP network. The network 12 could be implemented as a Voice Over FrameRelay or Voice Over ATM network, and the interfaces and protocolssupported by the test scheme could be modified accordingly.

What is claimed is:
 1. A method of providing service for use in a VoiceOver Internet Protocol (VOIP) network environment comprising: selectinga service level; and measuring voice call listening quality according tothe selected service level for voice calls transmitted across a VOIPnetwork to produce voice call listening quality metric values.
 2. Themethod of claim 1, wherein the selected service level is associated witha type of voice codec.
 3. The method of claim 2, wherein the type ofvoice codec comprises a waveform codec.
 4. The method of claim 1,wherein measuring comprises measuring the voice call listening qualityusing a perceptual test model.
 5. The method of claim 4, wherein theperceptual test model comprises Perceptual Analysis Measurement System(PAMS).
 6. The method of claim 4, wherein the perceptual test modelcomprises Perceptual Speech Quality Measurement (PSQM).
 7. The method ofclaim 2, wherein the type of voice codec comprises a hybrid codec. 8.The method of claim 1, wherein the voice call listening quality metricvalue corresponds to a Mean Opinion Score (MOS) value.
 9. The method ofclaim 1, further comprising: using the measured voice call listeningquality metric values to determine whether a service level agreementguarantee provided to a user of the VOIP network is met.
 10. The methodof claim 1, wherein measuring comprises: controlling test probesdeployed along the border of the VOIP network to engage each other intest calls and to make voice call listening quality measurements basedon the test calls.
 11. The method of claim 10, wherein the test probesare connected to VOIP communication devices that are connected to theVOIP network.
 12. The method of claim 11, wherein the VOIP communicationdevices comprise gateways.
 13. The method of claim 1, wherein measuringcomprises: controlling test probes deployed at edges of the VOIP networkto engage each other in test calls and to make voice call listeningquality measurements based on the test calls.
 14. The method of claim 1,wherein measuring comprises: controlling at least one test probedeployed at and connected to a telephony network that is coupled to theVOIP network by a gateway to generate test voice calls and to make voicecall listening quality measurements based on the generated test voicecalls.
 15. The method of claim 2, wherein the selected service level isfurther associated with a VOIP signaling protocol.
 16. The method ofclaim 15, wherein the VOIP signaling protocol comprises H.323.
 17. Themethod of claim 15, wherein the VOIP signaling protocol comprises SIP.18. The method of claim 15, wherein the VOIP signaling protocolcomprises MGCP.
 19. A computer program product residing on a computerreadable medium for providing service for use in a Voice Over InternetProtocol (VOIP) network environment, comprising instructions for causinga computer to: associate service levels with phone numbers; andresponsive to a test voice call directed to one of the phone numbers,cause the test voice call to be transferred over the VOIP network to adestination corresponding to such phone number at the associated servicelevel and causing a voice call listening quality to be measured for theassociated service level to produce a voice call listening qualitymetric value.
 20. The computer program product of claim 19, wherein theservice levels correspond to different types of voice codecs.
 21. Thecomputer program product of claim 20, wherein the service levelscorrespond to different combinations of voice codec types and types ofVOIP signaling protocols.